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| 1 /* | 1 /* |
| 2 * libjingle | 2 * libjingle |
| 3 * Copyright 2004 Google Inc. | 3 * Copyright 2004 Google Inc. |
| 4 * | 4 * |
| 5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
| 6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
| 7 * | 7 * |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
| 9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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| 131 | 131 |
| 132 VoEWrapper* voe() { return voe_wrapper_.get(); } | 132 VoEWrapper* voe() { return voe_wrapper_.get(); } |
| 133 int GetLastEngineError(); | 133 int GetLastEngineError(); |
| 134 | 134 |
| 135 // Set the external ADM. This can only be called before Init. | 135 // Set the external ADM. This can only be called before Init. |
| 136 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); | 136 bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm); |
| 137 | 137 |
| 138 // Starts AEC dump using existing file. | 138 // Starts AEC dump using existing file. |
| 139 bool StartAecDump(rtc::PlatformFile file); | 139 bool StartAecDump(rtc::PlatformFile file); |
| 140 | 140 |
| 141 // Check whether the supplied trace should be ignored. | |
| 142 bool ShouldIgnoreTrace(const std::string& trace); | |
| 143 | |
| 144 // Create a VoiceEngine Channel. | 141 // Create a VoiceEngine Channel. |
| 145 int CreateMediaVoiceChannel(); | 142 int CreateMediaVoiceChannel(); |
| 146 | 143 |
| 147 private: | 144 private: |
| 148 typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; | 145 typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList; |
| 149 typedef sigslot:: | 146 typedef sigslot:: |
| 150 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; | 147 signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal; |
| 151 | 148 |
| 152 void Construct(); | 149 void Construct(); |
| 153 void ConstructCodecs(); | 150 void ConstructCodecs(); |
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| 436 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 433 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 437 | 434 |
| 438 // Do not lock this on the VoE media processor thread; potential for deadlock | 435 // Do not lock this on the VoE media processor thread; potential for deadlock |
| 439 // exists. | 436 // exists. |
| 440 mutable rtc::CriticalSection receive_channels_cs_; | 437 mutable rtc::CriticalSection receive_channels_cs_; |
| 441 }; | 438 }; |
| 442 | 439 |
| 443 } // namespace cricket | 440 } // namespace cricket |
| 444 | 441 |
| 445 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ | 442 #endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_ |
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